Add constructor options documentation

Xander Dumaine · Xander Dumaine · commit 64cf6b2ee130 · 2015-12-24T16:53:08.000-05:00
diff --git a/README.md b/README.md
@@ -14,7 +14,7 @@ Want to see it in action? Check out the demo: https://talky.io/
 <!DOCTYPE html>
 <html>
     <head>
-        <script src="http://simplewebrtc.com/latest-v2.js"></script> 
+        <script src="http://simplewebrtc.com/latest-v2.js"></script>
         <style>
             #remoteVideos video {
                 height: 150px;
@@ -60,7 +60,7 @@ webrtc.on('readyToCall', function () {
 
 `peerConnectionConfig` - Set this to specify your own STUN and TURN servers. By default, SimpleWebRTC uses Google's public STUN server (`stun.l.google.com:19302`), which is intended for public use according to: https://twitter.com/HenrikJoreteg/status/354105684591251456
 
-Note that you will most likely also need to run your own TURN servers. See http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/ for a basic tutorial. 
+Note that you will most likely also need to run your own TURN servers. See http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/ for a basic tutorial.
 
 ## Filetransfer
 Sending files between individual participants is supported. See http://simplewebrtc.com/filetransfer.html for a demo.
@@ -73,7 +73,7 @@ Sometimes you need to do more advanced stuff. See http://simplewebrtc.com/notsos
 
 ## Got questions?
 
-Join the SimpleWebRTC discussion list: 
+Join the SimpleWebRTC discussion list:
 
 http://lists.andyet.com/mailman/listinfo/simplewebrtc
 
@@ -82,3 +82,102 @@ or the Gitter channel:
 https://gitter.im/HenrikJoreteg/SimpleWebRTC
 
 ## API
+
+### Constructor
+
+`new SimpleWebRTC(options)`
+
+- `object options` - options object provided to constructor consisting of:
+  - `string url` - *required* url for signaling server. Defaults to signaling server URL which can be used for development. You must use your own signaling server for production.
+  - `object sockio` - *optional* object to be passed as options to the signaling server connection.
+  - `Connection connection` - *optional* connection object for signaling. See `Connection` below. Defaults to a new SocketIoConnection
+  - `bool debug` - *optional* flag to set the instance to debug mode
+  - `[string|DomElement] locaVidelEl` - ID or Element to contain the local video element
+  - `[string|DomElement] remoteVideosEl` - ID or Element to contain the
+  remote video elements
+  - `bool autoRequestMedia` - *optional(=false)* option to automatically request user media. Use `true` to request automatically, or `false` to request media later with `startLocalVideo`
+  - `bool enableDataChannels` *optional(=true)* option to enable/disable data channels (used for volume levels or direct messaging)
+  - `bool autoRemoveVideos` - *optional(=true)* option to automatically remove video elements when streams are stopped.
+  - `bool adjustPeerVolume` - *optional(=false)* option to reduce peer volume when the local participant is speaking
+  - `number peerVolumeWhenSpeaking` - *optional(=.0.25)* value used in
+  conjunction with `adjustPeerVolume`. Uses values between 0 and 1.
+  - `object media` - media options to be passed to `getUserMedia`. Defaults to `{ video: true, audio: true }`. Valid configurations described [on MDN](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia) with official spec [at w3c](http://w3c.github.io/mediacapture-main/#dom-mediadevices-getusermedia).
+  - `object receiveMedia` - *optional* RTCPeerConnection options. Defaults to `{ offerToReceiveAudio: 1, offerToReceiveVideo: 1 }`.
+  - `object localVideo` - *optional* options for attaching the local video stream to the page. Defaults to
+  ```javascript
+  {
+      autoplay: true, // automatically play the video stream on the page
+      mirror: true, // flip the local video to mirror mode (for UX)
+      muted: true // mute local video stream to prevent echo
+  }
+  ```
+  - `object logger` - *optional* alternate logger for the instance; any object that implements `log`, `warn`, and `error` methods.
+
+### Fields
+
+`capabilities` - the [`webrtcSupport`](https://github.com/HenrikJoreteg/webrtcsupport) object that describes browser capabilities, for convenience
+
+`config` - the configuration options extended from options passed to the constructor
+
+`connection` - the socket (or alternate) signaling connection
+
+`webrtc` - the underlying WebRTC session manager
+
+### Events
+
+`'connectionReady', sessionId`
+
+`'createdPeer', peer`
+
+`'stunservers', stunServers`
+
+`'turnservers', turnservers`
+
+`'localScreenAdded', el`
+
+### Methods
+
+`leaveRoom()`
+
+`disconnect()`
+
+`handlePeerStreamAdded(peer)`
+
+`handlePeerStreamRemoved(peer)`
+
+`getDomId(peer)`
+
+`setVolumeForAll(volume)`
+
+`joinRoom(name, callback)`
+
+`getEl(idOrEl)`
+
+`startLocalVideo()`
+
+`stopLocalVideo()`
+
+`getLocalVideoContainer()`
+
+`getRemoteVideoContainer()`
+
+`shareScreen(callback)`
+
+`getLocalScreen()`
+
+`stopScreenShare()`
+
+`testReadiness()`
+
+`createRoom(name, callback)`
+
+`sendFile()`
+
+### Connection
+
+By default, SimpleWebRTC uses a Socket.io connection to communicate with the signaling server. However, you can provide an alternate connection object to use. All that your alternate connection need provide are four methods:
+
+- `on(ev, fn)` - A method to invoke `fn` when event `ev` is triggered
+- `emit()` - A method to send/emit arbitrary arguments on the connection
+- `getSessionId()` - A method to get a unique session Id for the connection
+- `disconnect()` - A method to disconnect the connection
